[asterisk-users] Asterisk DIAL() premature timeout on a PRI trunk to legacy PBX

[Tzafrir Cohen]

On Sun, Sep 16, 2007 at 03:45:15AM -0700, Vieri wrote:
> 
> --- Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
> 
> > You can probably get an answer to that if you enable
> > and log debug
> > messages of Asterisk .
> 
> Thanks, I thought that a pri debug was enough but now
> I have the missing information:
> 
> Sep 16 12:37:28 VERBOSE[19175] logger.c:     --
> Zap/1-1 is ringing
> Sep 16 12:37:32 DEBUG[9060] chan_sip.c: Auto
> destroying call
> '1457599B12DC407D91479E2D1EEB51960xc0a8fe05'
> Sep 16 12:37:46 DEBUG[19175] dsp.c: ast_dsp_busydetect
> detected busy, avgtone: 1525, avgsilence 3000
> Sep 16 12:37:46 DEBUG[19175] dsp.c: Requesting Hangup
> because the busy tone was detected on channel Zap/1-1
> Sep 16 12:37:46 VERBOSE[19175] logger.c:     --
> Zap/1-1 is busy
> 
> So I guess it's clear enough: the Alcatel is the first
> party to force the connection to be dropped because it
> issues a busy tone. Is that right?
> 
> Can I honestly say that it's an Alcatel issue and that
> nothing can be done on the Asterisk side?

Hmmm.... I wasn't reading this properly....

I guess it is. Though I wonder why it doesn't use proper signalling.

-- 
               Tzafrir Cohen       
icq#16849755                    jabber:tzafrir at jabber.org
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http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir